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Summary:ASTERISK-14818: asterisk does not send by "BYE" to sip peer
Reporter:Stefan Baldus (stefanero)Labels:
Date Opened:2009-09-11 03:29:44Date Closed:2011-06-07 14:00:26
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) asterisk-sip-debug.txt
( 1) nortel-channel-output.txt
( 2) wireshar-pic-trace.jpg
Description:Hello everybody,

I have a problem regarding not closing sip sessions.
Let me describe first our setup we have.


FAX <--analog line--> Linksys ATA (spa2102) <--SIP--> astersik (1.6.1.6) <--SIP--> Nortel-CS1KE (rel 5.5) <--ISDN--> external FAX


When I start to send a fax over the ATA to an external fax we noticed that asterisk does not send a BYE msg to the CS1KE. Because of that the line to the external FAX stays "open" and so the tariff rate is still counting.

The ATA sends a BYE to the asterisk and so CDR records are written, but asterisk does not send the BYE to Nortel.

I will attach a wireshark trace, a sip debug from asterisk and a information screen from the nortel regarding the channel.


Regards
Stefanero

****** ADDITIONAL INFORMATION ******

OS: openSuSE 11.1
* version: 1.6.1.6 / 1.6.0.15 (happens with both)

Used IP:
*.*.*.10   asterisk server
*.*.*.139 linksys ata
*.*.*.51 SIG Server Nortel
*.*.*.143 Media GW Nortel
*.*.*.200 Nortel Core

I left most settings unchanged in sip.conf from default 1.6.1.6 install

relevant sip settings changes:
t38pt_udptl = yes

[cs1000e]
type=peer
host=*.*.*.200
usereqphone=yes
context=callout

[nst1012]
username=nst1012
type=friend
secret=******
host=dynamic
callerid=nst1012 <1012>
context=callout
Comments:By: Stefan Baldus (stefanero) 2009-09-11 03:47:49

Sorry I am not allowed to upload the wireshark file, only as jpeg image.
Hope you can still see in the end, that the ATA sends teh bye to the asterisk server, but asterisk never passes it on to the nortel.
By: Paul Belanger (pabelanger) 2010-06-02 13:32:03

Please retest with 1.6.2
--
Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.6.0 and 1.6.1 branches has ended. For continued maintenance support please move to the 1.6.2 branch.

More information on this change can be found in the release announcement: http://www.asterisk.org/node/49924

By: Paul Belanger (pabelanger) 2010-06-10 15:08:46

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.

Further information can be found at http://www.asterisk.org/developers/bug-guidelines